Bandwidth optimiser user guide.

Home dashboard

The service panel home screen summarises live server health, capacity and call quality at a glance:

• Server clock & info: Displays configured timezone, dedicated server IP, allocated port limit and continuous uptime.
• Current calls: Real-time counters for Total, Active (connected) and Ringing channels.
• Quality statistics: Today’s ASR, ACD and total billed minutes — plus quick-stat lookbacks for the last 3 hours, 24 hours and 7 days.
• Client tunnel status: Shape-and-status badges indicate Up (client reachable and paired), Pending (client offline or booting), or Interrupted (the tunnel is not getting through — try another tunnel mode). Always sign out when finished managing your server.

Carrier & codec payload control

Register originating carrier switches over SIP (UDP port 5060) and restrict or tune allowed audio codecs and packetisation intervals (ms) in the Allowed codecs field:

• Default profile permits g729:20 and g723:30 (codec name followed by payload size in milliseconds).
• To allow all supported codecs (g711a, g711u, g729, g723), enter all.
• Higher packetisation intervals (for example g729:40 or g723:60) further reduce packet-header overhead where carrier jitter tolerances permit.

SIP user creation & test calls

Create SIP credentials directly inside the panel to place end-to-end test calls from a SIP softphone or register a remote gateway over SIP:

1. Open SIP user, enter a username, strong password, concurrent call limit and codec, then select Save now.
2. For quick mobile testing, temporarily disable SIP firewall under Utility (remember to re-enable it for production carrier traffic) and reload settings.
3. Configure your cloud server IP, SIP username and password in any standard SIP softphone or endpoint to verify two-way audio.

Cloud client tunnels & router TFTP firmware recovery

Add the hardware MAC address of your Cloud PC, Cloud Pi or Cloud router under Cloud client. On PC and Pi deployments, you can switch between SX, TX (multi-port range including port 5967) and UX tunnel modes to find the transport that works on your link; Cloud router firmware includes built-in CMER tunnelling.

Flashing router firmware (web UI & TFTP fallback)

1. Attempt a standard upgrade first via your router’s firmware upgrade page using the matching CMER Router OS v1.2 binary, then connect the internet feed to the WAN port and wait 5–10 minutes for the client status to turn active.
2. If the stock web UI rejects the image, install a TFTP server on a Windows PC wired to the router LAN port and set a static IPv4 address of 192.168.0.66 (subnet 255.255.255.0, gateway 192.168.0.1).
3. Rename the recovery firmware file to match your model’s recovery filename (for example mr3420v3_tp_recovery.bin), point TFTP to that folder and select interface 192.168.0.66.
4. Power off the router, hold the Reset button while powering on for 4–5 seconds, then release. TFTP transfers the firmware automatically and reboots the router into CMER Router OS v1.2.

Gateway management & remote gateway browsing

Register LAN gateways (192.168.101.x) or public-IP endpoints under Gateway. When you cannot reach a gateway with your usual remote-access tool, toggle the gateway browsing action icon to open a secure reverse proxy directly to your gateway’s web admin UI.

Dial plan routing & digit-length protection

• Null mod-prefix stripping: Leave Mod prefix blank to strip country/trunk prefixes cleanly (for example stripping 967 where no leading national zero is used).
• Special bundle prefixes: Prefix and mod-prefix fields accept _, #, $, * and + for operator SIM bundle codes (such as *123#0).
• Priority failover: Routes evaluate in ascending order (0 first, failing over to 1, 2, 3…).
• Allowed length filter: Specify exact expected digit counts (e.g. 15) to drop malformed or junk dial strings before they reach SIM ports.

Profile & timezone settings

Maintain your administrator email, display name, panel password and timezone under Profile. Updating your timezone automatically adjusts all CDR reports and scheduled restart timers across the panel.

Utility controls & debug console

• Retry dial (default off): Automatically re-attempts failed first-ring calls once to lift ASR on congested routes.
• Early ring (default off): Sends immediate SIP 183 session progress ringback to reduce perceived post-dial delay (PDD).
• Dial timeout (default on): Controls SIP 603 timeout signalling behaviour.
• SIP firewall (default on): Restricts inbound SIP traffic strictly to IP addresses whitelisted in your carrier table.
• Always browsing (default off): Keeps gateway browsing open continuously instead of auto-closing after 30 minutes.
• Route to SIP user (default off): Enables inbound call termination directly to registered SIP user accounts.
• Debug console: Live SIP and channel trace viewer showing originating switch IPs and negotiated codecs in real time.

Active calls monitor

Inspect live sessions with carrier, gateway, caller ID, called number, status (Active / Ringing) and elapsed duration. Sort any column header, filter by number substring, adjust page size (10 / 25 / 50 / 100), or terminate an individual hung channel instantly.

Call detail records (CDRs) & PDF export

Query historical traffic by custom date range in Daily summary or Hourly breakdown views. Filter by specific carrier, gateway or prefix, enter an optional per-second rate and currency to compute billing totals, and export formatted PDF reports in one click.

Automated restart scheduler

Configure unattended daily or weekly restarts for your cloud server service or paired client device during low-traffic maintenance windows, automatically aligned to your profile timezone.

VoIP terminology glossary

ACD (average call duration)

Mean billed length of answered calls over a selected interval — a primary indicator of route audio quality.

ASR (answer-seizure ratio)

Percentage of seized call attempts that successfully connect and answer.

CDR (call detail record)

Timestamped log of call origin, destination, disposition, duration and codec metadata.

CMER

Compressed, multiplexed & encrypted RTP — reduces G.729a voice bandwidth from ~42 kbps down to 8.1 kbps while encrypting media.

MAC address

12-character hexadecimal hardware identifier (0–9, A–F) of a router, Pi or PC network interface used to pair a cloud client.

PDD (post-dial delay)

Elapsed time between the final dialled digit and audible ringback or answer supervision.

Packet loss

Percentage of real-time RTP frames dropped in transit; voice links should remain well below 5% loss.